Voice over Internet Protocol (VoIP) is the technology of using digital signal transmission techniques to transmit voice communications in real-time and at least partially over the Internet. VoIP enables an audio or video call over Internet Protocol (IP) networks as a cost-effective alternative of voice transmission that exclusively uses traditional public switched telephone networks (PSTNs) by long distance exchange carriers. Due to the digital nature of the transmission, VoIP provides increased signal processing, encryption and call set-up and operating capabilities.
Typically, VoIP messages are transmitted using the Real-time Transport Protocol (RTP), the RTP Control Protocol (RTCP), and the Session Initiation Protocol (SIP). Accordingly, a VoIP system includes RTP server(s) and SIP server(s). The RTP defines a standardized packet format and carries the media streams (e.g., audio and video data), while the RTCP is used to monitor transmission statistics and Quality of Service (QoS) and aid synchronization of multiple streams. The SIP assists in setting up and ending connections of various system components across the network.
In existing VoIP systems, each user is associated with a user account that is independent of a telephone number, such as a landline phone number or a mobile phone number. Usually a user ID can be any string of characters chosen by a user. A caller-user and a callee-user must be both logged in their VoIP accounts at the same time to initiate a VoIP call. A major cause for frustration in using VoIP is that a user account (especially a callee) often stays logged out or offline so the caller cannot reach the callee, for example while the VoIP application program is closed on the callee's IP phone, the callee's IP phone is shut down or has no access to WiFi or any other network service, and etc. In contrast, a user's landline phone or cell phone typically remain standby or open constantly.